On Mon, 2008-10-20 at 14:01 -0500, Steve Anness wrote:
I am sure this has been discussed prior, however, I am sitting here
and being asked this very question by my superiors.
Ahh stuperiors, don't you love the questions they ask? Almost as good as
the questions some recruiters (by this I mean
hi there,
I 'm a newbie in VOIP technologies ; i 'm implementing asterisk and i 'm
wonder what is the best way to resolving the Asterisk/NAT problem : some
clients are behind a NAT.
anyone could help me?
thanks
johanna
_
hi there,
I 'm a newbie in VOIP technologies ; i 'm implementing asterisk and i 'm
wonder what is the best way to resolving the Asterisk/NAT problem : some
clients are behind a NAT.
anyone could help me?
thanks
johanna
_
John,
Client Behind a NAT should not be problem. What are your issues? If you post
your scenario and more details about your problem only then some can help
you better.
Jai
Buy SIP DID at www.didforsale.com
On Wed, Oct 22, 2008 at 12:24 AM, Johanna NIRINA [EMAIL PROTECTED]wrote:
hi there,
I
I'm using asterisk 1.4 . There is some sip clients is behind a NAT : the
asterisk server can't send request to these client. I'm looking for a solution
to solve that in the server (asterisk) side. (sorry for my english).
thanks,
johanna
On 20 Oct 2008, at 20:01, Steve Anness wrote:
I am sure this has been discussed prior, however, I am sitting here
and being asked this very question by my superiors. They are loving
what I have done with our two Asterisk servers here; however, they
keep asking me if it is secure or
On 22 Oct 2008, at 07:23, Nikolai Lusan wrote:
On Mon, 2008-10-20 at 14:01 -0500, Steve Anness wrote:
However, realistically if I am using the asterisk server to make
internal calls and discussion very private matters, how possible is
it
for someone to listen to calls? How good is the
On 22 Oct 2008, at 10:44, voip crazy wrote:
Hello list,
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Yep, take a look at our offering on www.phonefromhere.com
Tim.
Hello list,
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Thanks.
VoipCrazy
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asterisk-users
I am trying to setup a second asterisk box to play with console/dsp over
sip.
My sip.conf on the second box is:
[secondbox]
type=friend
username=secondbox
secret=secret
disallow=all
allow=ulaw
allow=alaw
allow=gsm
host=SERVERIP
context=consoledsp
The second box is not connecting to my asterisk
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne av Dwayne Hubbard
Sendt: 21. oktober 2008 19:04
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Asterisk Console color
- Armand Fumal
Tim Panton wrote:
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Yep, take a look at our offering on www.phonefromhere.com
A per-minute charge does not constitute a free solution. Please read
requests
On 22 Oct 2008, at 14:28, Rob Hillis wrote:
Tim Panton wrote:
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Yep, take a look at our offering on www.phonefromhere.com
A per-minute charge does not constitute
Jerry Geis wrote:
I am trying to setup a second asterisk box to play with console/dsp over
sip.
My sip.conf on the second box is:
[secondbox]
type=friend
username=secondbox
secret=secret
disallow=all
allow=ulaw
allow=alaw
allow=gsm
host=SERVERIP
context=consoledsp
The second box is not
Johanna NIRINA wrote:
I'm using asterisk 1.4 . There is some sip clients is behind a NAT : the
asterisk server can't send request to these client. I'm looking for a solution
to solve that in the server (asterisk) side. (sorry for my english).
thanks,
johanna
On Tue, Oct 21, 2008 at 06:43:31PM +0200, Armand Fumal wrote:
Hi,
Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color in
console.
Do I miss a package or compilation option ?
http://bugs.digium.com/9048
The fix for this issue is to tell Asterisk to pretend it has a valid
HI all,
I have a question, is call parking broken:
When you park a call it says it will time out to a certain extension in a
certain context, it never does it just calls the parker back.
How do you get it to timeout to certain extension?
-- Executing [EMAIL PROTECTED]:2]
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote:
Does anyone know what the significance is of the b1 being sent here?
Or, is there a way to make Asterisk not send the b1 character as a
test?
As an update to this, I noticed the following lines in libpri.h near
line 236:
/* Network
I'm trying to figure out how to handle our fax line when we switch to
our asterisk for voice. After a lot of reading and poking about I have
concluded, as have many others it would seem, that the best thing to
do is either to have a separate pstn fax line or use some sort of
internet
hi,
hs anyone able to make video to work on asterisk? i tried following this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam
i can see that eyebeam is trying to broadcast a video but the other
eyebeam is not receiving it.
i tested the same setup but this time using
Hi,
I'm having an issue where some phones behind a sonicwall are auto-congesting.
The status on sip show peer shows ping times anywhere from 80ms all
the way up to 1100ms.
PCs behind the same firewall have a ping time of about 30ms to the PBX itself.
Does anyone know if the sonicwall is inserting
hi
for any context ,you must to open /etc/asterisk/extensions.conf and insert
this line : exten =Realtime/[EMAIL PROTECTED]
and (reload) or (restart now) your asterisk
You don't have to restart asterisk, just a 'dialplan reload' will
suffice. So really there is no impact to a running
What version of *? Are you going all VOIP for your voice or are you using a
T1/E1? *?
1.4 has t38 pass-through and 1.6 has pass-through and termination, but 1.6 was
just release and I would not suggest using it in a production environment
unless you can tolerate problem or even outages.
If
Sonicwalls from the TZ line and before line do seem to have a number of
issues with VoIP.
Jeff Johnson
Director of Operations
NeturallySpeaking, LLC
sip://[EMAIL PROTECTED]
http://www.neturallyspeaking.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
I had weird issues when using a Sonicwall, gave up. Stuck in linksys running
dd-wrt firmware running on a separate VLAN... no issues since
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Lamanna
Sent: Wednesday, October 22, 2008 12:35 PM
To:
hi, using sip, my default codec is set to gsm in sip.conf
I occasionally want to send out a call using ulaw while other channels
are using gsm, how can I do this using call files ?
I couldn't find any codec parameter in the call file definition.
tia.
HI all,
I have a question, is call parking broken:
When you park a call it says it will time out to a certain extension in a
certain context, it never does it just calls the parker back.
How do you get it to timeout to certain extension?
-- Executing [EMAIL PROTECTED]:2]
On Wednesday 22 October 2008 12:27:17 Nhadie wrote:
hs anyone able to make video to work on asterisk? i tried following this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam
i can see that eyebeam is trying to broadcast a video but the other
eyebeam is not receiving
hi
for any context ,you must to open /etc/asterisk/extensions.conf and
insert this line : exten =Realtime/[EMAIL PROTECTED]
and (reload) or (restart now) your asterisk
You don't have to restart asterisk, just a 'dialplan reload' will
suffice. So really there is no impact to a running
Sonicwalls TZ170 and older have issues with SIP
Jeff Johnson
Director of Operations
NeturallySpeaking, LLC
sip://[EMAIL PROTECTED]
http://www.neturallyspeaking.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Van
Ham
Sent: Wednesday, October
On Wed, 2008-10-22 at 12:11 -0500, Bob Pierce wrote:
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote:
Does anyone know what the significance is of the b1 being sent here?
Or, is there a way to make Asterisk not send the b1 character as a
test?
As a further update to this, I've
On 22 Oct 2008, at 20:29, Craig Van Ham wrote:
HI all,
snip
This appears to be the same message you posted earlier.
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To UNSUBSCRIBE or update
I also tried downgrading to version 1.4-current but that didn't help.
Any other ideas? I'm at a loss.
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What kind of phone are you trying to connect to 101??? and from where?
On Wed, Oct 22, 2008 at 7:07 PM, Stephen Reese [EMAIL PROTECTED] wrote:
I also tried downgrading to version 1.4-current but that didn't help.
Any other ideas? I'm at a loss.
--
Juan E. Rodríguez
Cel. 829-886-5565
hi sir,
i uncommented that as mentioned on the howto.
regards,
nhadie
Tilghman Lesher wrote:
On Wednesday 22 October 2008 12:27:17 Nhadie wrote:
hs anyone able to make video to work on asterisk? i tried following this:
On Wed, Oct 22, 2008 at 8:15 PM, Juan Rodríguez [EMAIL PROTECTED] wrote:
What kind of phone are you trying to connect to 101??? and from where?
Both phones are Cisco, 101 is a 7960 and 102 is a 7912. 101 can
contact 102 by dialing 101 but not the other way around, I just get a
busy tone.
I hadn't seen anything on the asterisk list but just in case anyone is
interest.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: Murat Aktihanoglu [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 22
TDE does NOT need a VoIP card, you need to buy a DSP card, VoIP is
built in. In fact if you are making pure VoIP extensions to extension
calls you don't even need the DSP card/s.
What type of VoIP are you trying to accomplish with Asterisk?
Extensions to extensions? or Provider based.
In any event
On Tue, Oct 21, 2008 at 11:55 AM, Rodolfo Alcazar Portillo
[EMAIL PROTECTED] wrote:
Am Dienstag, den 21.10.2008, 06:54 -0600 schrieb César García:
Hello Rodolfo,
I see you have experience with Panasonic, and I have a new challenge
of integrating Asterisk in an enterprice where they have a
I am using 1.6.0.1 and we are going to be pure voip. I know it has
pass through and termination, but that is useless if I don't have a
way to transform the analog t.30 to t.38 before it gets to me. That is
where my confusion lays, is there some way of doing this that I am not
aware of?
If you are VoIP-only then you need a SIP provider that offers T.38.
On Wed, Oct 22, 2008 at 11:17 PM, Brendan Martens
[EMAIL PROTECTED] wrote:
I am using 1.6.0.1 and we are going to be pure voip. I know it has
pass through and termination, but that is useless if I don't have a
way to transform
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